Audio coding

ABSTRACT

An audio coder is arranged to process a respective set of sampled signal values for each of a plurality of sequential segments of an audio signal (x). The coder comprises an analyser (TSA) arranged to analyse the sampled signal values to provide one or more sinusoidal codes (Cs) corresponding to respective sinusoidal components of the audio signal. A subtractor subtracts a signal corresponding to the sinusoidal components from the audio signal to provide a first residual signal (r 1 ). A modeller (SEG) models the frequency spectrum of the first residual signal (r 1 ) by determining first filter parameters (Ps) of a filter which has a frequency response approximating a frequency spectrum of the first residual signal. Another subtractor subtracts a signal corresponding to the first filter parameters from the first residual signal to provide a second residual signal (r 2 ). Another modeller (RPE) models a component (r 2 ,r 3 ) of the second residual signal with a pulse train coder (RPE) to provide respective pulse train parameters (L 0 ). A bit stream generator ( 15 ) generates an encoded audio stream (AS) including the sinusoidal codes (Cs), the first filter parameters (Ps) and the pulse train parameters (L 0 ).

FIELD OF THE INVENTION

The present invention relates to coding and decoding audio signals.

BACKGROUND OF THE INVENTION

Referring now to FIG. 1, a parametric coding scheme in particular asinusoidal coder is described in US Published Application No.2001/0032087A1. In this coder, an input audio signal x(t) received froma channel 10 is split into several (overlapping) segments or frames,typically of length 20 ms. Each segment is decomposed into transient(C_(T)), sinusoidal (C_(S)) and noise (C_(N)) components. (It is alsopossible to derive other components of the input audio signal such asharmonic complexes although these are not relevant for the purposes ofthe present invention.)

The first stage of the coder comprises a transient coder 11 including atransient detector (TD) 110, a transient analyzer (TA) 111 and atransient synthesizer (TS) 112. The detector 110 estimates if there is atransient signal component and its position. This information is fed tothe transient analyzer 111. If the position of a transient signalcomponent is determined, the transient analyzer 111 tries to extract(the main part of) the transient signal component. It matches a shapefunction to a signal segment preferably starting at an estimated startposition, and determines content underneath the shape function, byemploying for example a (small) number of sinusoidal components. Thisinformation is contained in the transient code C_(T).

The transient code C_(T) is furnished to the transient synthesizer 112.The synthesized transient signal component is subtracted from the inputsignal x(t) in subtractor 16, resulting in a signal x₂.

The signal x₂ is furnished to a sinusoidal coder 13 where it is analyzedin a sinusoidal analyzer (SA)130, which determines the (deterministic)sinusoidal components. The end result of sinusoidal coding is asinusoidal code C_(S) and a more detailed example illustrating theconventional generation of an exemplary sinusoidal code C_(S) isprovided in PCT patent application No. WO00/79519A1.

From the sinusoidal code C_(S)generated with the sinusoidal coder, thesinusoidal signal component is reconstructed by a sinusoidal synthesizer(SS) 131. This signal is subtracted in subtractor 17 from the input x₂to the sinusoidal coder 13, resulting in a remaining signal x₃ devoid of(large) transient signal components and (main) deterministic sinusoidalcomponents.

The remaining signal x₃ is assumed to mainly comprise noise and a noiseanalyzer 14 produces the noise code C_(N) representative of this noise,as described in, for example, PCT patent application No. WO01/89086A1.

FIGS. 2(a) and (b) show generally the form of an encoder (NE) suitablefor use as the noise analyzer 14 of FIG. 1 and a corresponding decoder(ND) for use as the noise synthesizer 33 of FIG. 6 (described later). Afirst audio signal r₁, corresponding to the residual x₃ of FIG. 1,enters the noise encoder comprising a first linear prediction (SE) stagewhich spectrally flattens the signal and produces predictioncoefficients (Ps) of a given order. More generally, a Laguerre filtercan be used to provide frequency sensitive flattening of the signal asdisclosed in E. G. P. Schuijers, A. W. J. Oomen, A. C. den Brinker andA. J. Gerrits, “Advances in parametric coding for high-quality audio.”,Proc. 1st IEEE Benelux Workshop on Model based Processing and Coding ofAudio (MPCA-2002), Leuven, Belgium, 15 Nov. 2002, pp. 73-79. Theresidual r₂ enters a temporal envelope estimator (TE) producing a set ofparameters Pt and, possibly, a temporally flattened residual r₃. Theparameters Pt can be a set of gains describing the temporal envelope.Alternatively, they may be parameters derived from Linear Prediction inthe frequency domain such as Line Spectral Pairs (LSPs) or Line SpectralFrequencies (LSFs), describing a normalised temporal envelope, togetherwith a gain envelope.

In the parametric decoder (ND), a synthetic white noise sequence isgenerated (in WNG) resulting in a signal r₃′ with a temporally andspectrally flat envelope. A temporal envelope generator (TEG) adds thetemporal envelope on the basis of the received, quantised parametersP_(t)′ and a spectral envelope generator (SEG, a time-varying filter)adds the spectral envelope on the basis of the received, quantisedparameters P., resulting in a noise signal r₁′ corresponding to signaly_(n) of FIG. 6.

In a multiplexer 15, an audio stream AS is constituted which includesthe codes C_(T), C_(S) and C_(N).

The sinusoidal coder 13 and noise analyzer 14 are used for all or mostof the segments and amount to the largest part of the bit rate budget.

It is well known that parametric audio coders can give a fair to goodquality at relatively low bit rates for example 20 kbit/s. However, athigher bit rates the quality increase, as a function of increasing bitrate is rather low. Thus, an excessive bit rate is needed to obtainexcellent or transparent quality. It is therefore difficult to attaintransparency using parametric coding at bit rates comparable to thoseof, for example, waveform coders. This means that it is difficult toconstruct parametric audio coders having an excellent to transparentquality without an excessive usage of bit budget.

The reason for the fundamental difficulty in parametric coding reachingtransparency is in the objects that are defined. The parametric coder isvery efficient in encoding tonal components (sinusoids) and noisycomponents (noise coder). However, in real audio, a lot of signalcomponents fall into a grey area: they can neither be modelledaccurately by noise nor can they be modelled as (a small number of)sinusoids. Therefore, the very definition of objects in a parametricaudio coder, though very beneficial from a bit rate point of view formedium quality levels, is the bottleneck in reaching excellent ortransparent quality levels.

At the same time, traditional audio coders (sub-band and transform) giveexcellent to transparent coding quality at certain bit rates, typicallyin the order of 80-130 kbit/s for stereo signals sampled at 44.1 kHz.Combinations of transform and parametric coders (so-called hybridcoders) have been proposed for example as disclosed in European patentapplication no. 02077032.7 filed on May 24, 2002 (Attorney Docket No. ID609811/PHNL020478). Here spectro-temporal intervals of an audio signal,which would otherwise be sub-band coded, are selectively coded withnoise parameters in an attempt to reduce bit rate while maintainingaudio quality.

Alternatively, a transform or sub-band coder might be cascaded with aparametric coder of the type shown in FIG. 1. However, the expectedcoding gain for such an arrangement, where the parametric coder ispreceding the transform or sub-band coder, is minimal. This because theperceptually most important regions of the audio signal would becaptured by the sinusoidal coder, leaving little possibility for codinggain in the transform/sub-band coder.

Audio coders using spectral flattening and residual signal modellingusing a small number of bits per sample are disclosed in A. Harma andU.K. Laine, “Warped low-delay CELP for wide-band audio coding”, Proc.AES 17th Int. Conf.: High Quality Audio Coding, pages 207-215, Florence,Italy, 2-5 Sep, 1999; S. Singhal, “High quality audio coding usingmulti-pulse LPC”, Proc. 1990 Int. Conf. Acoustic Speech Signal Process.(ICASSP90), pages 1101-1104, Atlanta Ga., 1990, IEEE Picataway, N.J.;and X. Lin, “High quality audio coding using analysis-by synthesistechnique”, Proc. 1991 Int. Conf. Acoustic Speech Signal Process.(ICASSP91), pages 3617-3620, Atlanta Ga., 1991, IEEE Picataway, N.J. Ina number of studies, it has been shown that this coding strategy enablesan excellent to transparent quality at bit rates corresponding to 2bit/sample for mono signals (88.2 kbit/s for 44.1 kHz audio). In thatrespect, they do not exceed the performance of sub-band or transformcoders.

It is an object of the present invention to provide a parametric audiocoder whose bit rate is controllable across a range and which provideshigh quality levels at a bit rate comparable with traditional coders.

DISCLOSURE OF THE INVENTION

According to the present invention, there is provided a method accordingto claim 1.

The invention provides scalability in a parametric coder, bysupplementing the noise coder with a pulse train coder. This provides alarge range of bit rate operating points and merges the two strategiesinto one coder without introducing a large overhead in complexity.

The coding strategies within the noise coder are complementary in termsof strengths and weaknesses. The Linear Predictor in the pulse traincoder, for example, is inefficient in describing a tonal audio segment,but the sinusoidal coder can do this efficiently. Thus, for tonal itemslike harpsichord, the pulse train coder is unable to deliver transparentquality for a coarse quantisation of the residual. For other signals,the prediction order of the pulse train coder linear prediction stagehas to be very high to allow a coarse quantisation of the residual. Fornoise like signals, decimation of the residual signal is a problem andleads to a loss of brightness.

In the preferred embodiment, the coding strategies are combined to forma base layer using the parametric coder and an additional (bit ratecontrolled) pulse train layer. The bit rate resources required for thecombined techniques are less than the bit rate requirements pertechnique since both methods apply spectral flattening and,consequently, the bits needed for this stage only have to be investedonce. With the preferred embodiment, a bit rate range from 20-120 kbit/s(for stereo signals) can be covered with performance better than orcomparable with that of state-of-the-art coders.

BRIEF DESCRIPTION OF THE DRAWINGS

Embodiments of the invention will now be described, by way of example,with reference to the accompanying drawings, in which:

FIG. 1 shows a conventional parametric coder;

FIGS. 2(a) and (b) show a conventional parametric noise encoder (NE) andcorresponding noise decoder (ND) respectively;.

FIG. 3 shows an overview of a mono encoder according to a preferredembodiment of the present invention;

FIG. 4 shows an overview of a mono decoder according to a firstembodiment of the present invention; and

FIG. 5 shows an overview of a mono decoder according to a secondembodiment of the present invention.

DESCRIPTION OF THE PREFERRED EMBODIMENTS

In the preferred embodiment, a parametric audio coder of the type shownin FIG. 1 is supplemented with a pulse train coder of the type describedin P. Kroon, E. F. Deprettere and R. J. Sluijter, “Regular PulseExcitation—A novel approach to effective and efficient multipulse codingof speech”, IEEE Trans. Acoust. Speech, Signal Process, 34, 1986.Nonetheless, it will be seen that while the embodiment is described interms of a Regular Pulse Excitation (RPE) coder, the invention canequally be implemented with Multi-Pulse Excitation (MPE) techniques asdisclosed in U.S. Pat. No. 4,932,061 or an ACELP coder as described K.Jarvinen, J. Vainio, P. Kapanen, T. Honkanen, P. Haavisto, R. Salami, C.Laflamme, J-P. Adoul, “GSM enhanced full rate speech codec”, Proc.ICASSP-97, Munich (Germany), 21-24 Apr. 1997, Volume 2, pp. 771-774,each of which include a first LP based spectrally flattening stage.

In the preferred embodiment, an overall bit rate budget determinedaccording to the quality required from the coder, is divided into abit-rate B usable by the parametric coder and an RPE coding budget whichis inversely proportional to an RPE decimation factor D.

Referring now to FIG. 3, an input audio signal x is first processedwithin block TSA, (Transient and Sinusoidal Analysis) corresponding withblocks 11 and 13 of the parametric coder of FIG. 1. Thus, this blockgenerates the associated parameters for transients and noise asdescribed in FIG. 1. Given the bit rate B, a block BRC (Bit RateControl) preferably limits the number of sinusoids and preferablypreserves transients such that the overall bit rate for sinusoids andtransients is at most equal to B, typically set at around 20 kbit/s.

A waveform is generated by block TSS (Transient and SinusoidalSynthesiser) corresponding to blocks 112 and 131 of FIG. 1 using thetransient and sinusoidal parameters (C_(T) and C_(S)) generated by blockTSA and modified by the block BRC. This signal is subtracted from inputsignal x, resulting in signal r₁ corresponding to residual x₃ in FIG. 1.In general, signal r₁ does not contain sinusoids and transients.

From signal r₁, the spectral envelope is estimated and removed in theblock (SE) using a Linear Prediction or a Laguerre filter as in theprior art FIG. 2(a). The prediction coefficients Ps of the chosen filterare written to a bitstream AS for transmittal to a decoder as part ofthe conventional type noise codes C_(N). Then the temporal envelope isremoved in the block (TE) generating, for example, Line Spectral Pairs(LSP) or Line Spectral Frequencies (LSF) coefficients together with again, again as described in the prior art FIG. 2(a). In any case, theresulting coefficients Pt from the temporal flattening are written tothe bitstream AS for transmittal to the decoder as part of theconventional type noise codes C_(N). Typically, the coefficients Ps andPT require a bit rate budget of 4-5 kbit/s.

Because pulse train coders employ a first spectral flattening stage, theRPE coder can be selectively applied on the spectrally flattened signalr₂ produced by the block SE according to whether a bit rate budget hasbeen allocated to the RPE coder. In an alternative embodiment, indicatedby the dashed line, the RPE coder is applied to the spectrally andtemporally flattened signal r₃ produced by the block TE.

As is known from the documents referred to in the background, the RPEcoder performs a search in an analysis-by-synthesis manner on theresidual signal r₂/r₃. Given a decimation factor D, the RPE searchprocedure results in an offset (value between 0 and D-1), the amplitudesof the RPE pulses (for example, ternary pulses with values −1, 0 and 1)and a gain parameter. This information is stored in a layer Lo includedin the audio stream AS for transmittal to the decoder by a multiplexer(MUX) when RPE coding is employed.

Typically, the RPE coder require a bit rate of at least 40 kbit/s or soand is therefore switched on as the quality requirement and so bitbudget of the encoder is increased towards the higher end of the qualityrange. For the lower part of the quality range where the RPE coder isinitially employed, the bit rate B is decreased to less than the maximumbit rate allowed for when the parametric coder is employed alone. Thisenables a monotonically increasing overall bit rate budget range to bespecified for the coder with quality increasing in proportion to thebudget.

Experiments showed that the RPE coder results in a loss in brightness inthe reconstructed signal, especially when using high decimation factors(e.g. D=8). Adding some low-level noise to the RPE sequence mitigatesthis problem. In order to determine the level of the noise, a gain (g)is calculated on basis of, for example, the energy/power differencebetween a signal generated from the coded RPE sequence and residualsignal r₂/r₃. This gain is also transmitted to the decoder as part ofthe layer L₀ information.

Referring now to FIG. 4, a first embodiment of the decoder compatiblewith the embodiment of FIG. 1 where the RPE block processes the residualsignal r₂ is shown. A de-multiplexer (DeM) reads an incoming audiostream AS′ and provides the sinusoidal, transient and noise codes (Cs,C_(T) and C_(N)(Ps,P_(T))) to respective synthesizers SiS, TrS andTEG/SEG as in the prior art. As in the prior art, a white noisegenerator (WNG) supplies an input signal for the temporal envelopegenerator TEG. In the embodiment, where the information is available, apulse train generator (PTG) generates a pulse train from layer L₀ andthis is mixed in block Mx to provide an excitation signal r₂′. It willbe seen from the encoder, that as the noise codes C_(N)(Ps,P_(T)) andlayer L₀ were generated independently from the same residual r₂, thesignals they generate need to be gain modified to provide the correctenergy level for the synthesized excitation signal r₂′. In thisembodiment, in a mixer (Mx), the signals produced by the blocks TEG andPTG are frequency weighted, so that for low frequencies, most of thesignal r₂′ is derived from the pulse coded information L₀ and for highfrequencies most of the signal r₂′ is derived from the synthesized noisesource WNG/TEG.

The excitation signal r₂′ is then fed to a spectral envelope generator(SEG) which according to the codes Ps produces a synthesized noisesignal r₁′. This signal is added to the synthesized signals produced bythe conventional transient and sinusoidal synthesizers to produce theoutput signal {circumflex over (x)}.

In an alternative embodiment, the signal generated by the pulse traingenerator PTG is used instead of the signal generated by WNG as an inputto the temporal envelope generator as indicated by the hashed line.

Referring now to FIG. 5, a second embodiment of the decoder correspondswith the embodiment of FIG. 1 where the RPE block processes the residualsignal r₃. Here, the signal generated by a white noise generator (WNG)and processed by a block We, based on the gain (g) determined by thecoder; and the pulse train generated by the pulse train generator (PTG)are added to construct an excitation signal r₃′. Where layer L₀information is available, within block We, the noise sequence ishigh-pass filtered to remove the low frequencies, which perceptuallydegrade the reconstructed excitation signal—as in the first embodimentof the decoder, these components of the synthesized noise signal arebased on the output of the pulse train generator rather than the noisebased excitation signal. Of course, where layer Lo information is notavailable, the white noise is fed through the block We to be provided asthe excitation signal r₃′ to a temporal envelope generator block (TEG).

The temporal envelope coefficients (P_(T)) are then imposed on theexcitation signal r₃′ by the block TEG to provide the synthesized signalr₂′ which is processed as before. As mentioned above, this isadvantageous because a pulse train excitation typically gives rise tosome loss in brightness which, with a properly weighted additional noisesequence, can be counteracted. The weighting can comprise simpleamplitude or spectral shaping each based on the gain factor g.

As before, the signal is filtered by, for example, a Laguerre filter inblock SEG (Spectral Envelope Generator), which adds a spectral envelopeto the signal. The resulting signal is then added to the synthesizedsinusoidal and transient signal as before.

It will be seen that in either FIG. 4 or FIG. 5, if no PTG is beingused, the decoding scheme resembles the conventional sinusoidal coderusing a noise coder only. If the PTG is used, a RPE sequence is added,which enhances the reconstructed signal i.e. provides a higher audioquality.

It should be noted that in the embodiment of FIG. 5, in contrast to thestandard pulse coder (RPE or MPE), where a gain which is fixed for acomplete frame is used, a temporal envelope is incorporated in thesignal r₂′. By using such a temporal envelope, a better sound qualitycan be obtained, because of the higher flexibility in the gain profilecompared to a fixed gain per frame.

1. A method of encoding an audio signal (x), the method comprising, foreach of a plurality of segments of the signal, the steps of: analysing(TSA) the sampled signal values to provide one or more sinusoidal codes(Cs) corresponding to respective sinusoidal components of the audiosignal; subtracting a signal corresponding to said sinusoidal componentsfrom said audio signal to provide a first residual signal (r₁);modelling (SE) the frequency spectrum of the first residual signal (r₁)by determining first filter parameters (Ps) of a filter which has afrequency response approximating a frequency spectrum of the firstresidual signal; subtracting a signal corresponding to said first filterparameters from the first residual signal to provide a second residualsignal (r₂); modelling (RPE) a component (r₂, r₃) of the second residualsignal with a pulse train coder (RPE) to provide respective pulse trainparameters (L₀); and generating (15) an encoded audio stream (AS)including said sinusoidal codes (Cs), said first filter parameters (Ps)and said pulse train parameters (L₀).
 2. A method as claimed in claim 1further comprising the steps of: modelling (TE) the temporal envelope ofeach second residual signal by determining second parameters (P_(t)),and providing a third residual signal (r₃) by removing from the secondresidual signal the temporal envelope corresponding to said secondparameters; wherein said component of the second residual signalcomprises a respective third residual signal (r₃) and wherein saidgenerating step includes said second parameters in said encoded audiostream (AS).
 3. A method as claimed in claim 1 further comprising thestep of: modelling (TEG) the temporal envelope of the second residualsignal by determining second parameters (P_(T)), and wherein saidcomponent of each second residual signal comprises said second residualsignal (r₂); and wherein said generating step includes said secondparameters in said encoded audio stream (AS).
 4. A method as claimed inclaim 2 further comprising the step of: estimating a difference betweena signal corresponding to said pulse train parameters and said component(r₂, r₃) of each second residual signal; and wherein said generatingstep includes an indicator of said difference (g) in said encoded audiostream (AS).
 5. A method as claimed in claim 1 wherein said pulse traincoder is one of a regular pulse excitation (RPE) coder; a multiple-pulseexcitation (MPE) coder; or an ACELP coder.
 6. A method as claimed inclaim 1 wherein said first filter parameters (Ps) comprise one of:Laguerre or Linear Prediction filter parameters.
 7. A method as claimedin claim 2 wherein said second parameters (P_(T)) comprise one of:Linear Prediction parameters or Line Spectral Pairs (LSP) or LineSpectral Frequencies (LSF) coefficients together with respective gains.8. A method as claimed in claim 1 wherein said method comprises the stepof: estimating (TSA) a position of a transient signal component in theaudio signal; matching a shape function having shape parameters and aposition parameter to said transient signal; and including (15) theposition and shape parameters describing the shape function in saidaudio stream (AS).
 9. A method as claimed in claim 1 wherein the numberof said sinusoidal components is limited by a first bit rate budget (B),wherein said pulse train coder is limited to producing said pulse trainparameters (L₀) within a second bit rate budget, and wherein the sum ofsaid first and second bit rate budgets is selected from a rangeaccording to a required quality of encoding.
 10. Method of decoding anaudio stream, the method comprising the steps of: reading (DeM) anencoded audio stream (AS′) including, for each of a plurality ofsegments of an audio signal: sinusoidal codes (CS), pulse trainparameters (L₀), and first filter parameters (Ps); and employing (SiS)said sinusoidal codes to synthesize respective sinusoidal components ofthe audio signal; employing (PTG) said pulse train parameters (L₀) togenerate an excitation signal; imposing (SEG) a spectral envelopeaccording to said first filter parameters (Ps) on a first signal (r₂′) acomponent of which comprises said excitation signal, and adding saidsynthesized sinusoidal components and said spectrally filtered signal toproduce a synthesized audio signal ({circumflex over (x)}).
 11. A methodaccording to claim 10 wherein said encoded audio stream includes secondparameters (P_(T)), said method comprising the step of: imposing (TEG) atemporal envelope according to said second filter parameters (P_(T)) ona second signal (r₃′) a component of which comprises said excitationsignal, and wherein said first signal comprises said temporally filteredsignal (r₂′).
 12. A method according to claim 11 further comprising thesteps of: generating (WNG) a white noise signal; and adding said whitenoise signal to said excitation signal to provide said second signal(r₃′).
 13. A method according to claim 12 further comprising: high-passfiltering (We) said white noise signal.
 14. A method according to claim12 wherein a gain (g) to be applied to said white noise signal is readfrom said audio stream.
 15. A method according to claim 10 wherein saidencoded audio stream includes second filter parameters (P_(T)), themethod comprising the step of: imposing (TEG) a time domain envelopeaccording to said second filter parameters (Ps) on said excitationsignal, and wherein said spectral envelope is imposed on said temporallyfiltered signal (r₂′).
 16. A method according to claim 10 wherein saidencoded audio stream includes second filter parameters (P_(T)), themethod comprising the steps of: generating (WNG) a white noise signal;imposing (TEG) a time domain envelope according to said second filterparameters (Ps) on the white noise signal, and mixing said temporallyfiltered white noise signal with said excitation signal to provide saidsecond signal (r₂′); wherein said spectral envelope is imposed on saidsecond signal (r₂′).
 17. A method according to claim 16 wherein saidmixing step comprises spectrally weighting said temporally filteredwhite noise signal and said excitation signal.
 18. Audio coder arrangedto process a respective set of sampled signal values for each of aplurality of sequential segments of an audio signal (x), said codercomprising: an analyser (TSA) arranged to analyse the sampled signalvalues to provide one or more sinusoidal codes (Cs) corresponding torespective sinusoidal components of the audio signal; a subtractorarranged to subtract a signal corresponding to said sinusoidalcomponents from said audio signal to provide a first residual signal(r₁); a modeller (SEG) arranged to model the frequency spectrum of thefirst residual signal (r₁) by determining first filter parameters (Ps)of a filter which has a frequency response approximating a frequencyspectrum of the first residual signal; a subtractor arranged to subtracta signal corresponding to said first filter parameters from the firstresidual signal to provide a second residual signal (r₂); a modeller(RPE) arranged to model a component (r₂,r₃) of the second residualsignal with a pulse train coder (RPE) to provide respective pulse trainparameters (L₀); and a bit stream generator (15) for generating anencoded audio stream (AS) including said sinusoidal codes (Cs), saidfirst filter parameters (Ps) and said pulse train parameters (L₀). 19.Audio player, comprising: means for reading (DeM) an encoded audiostream (AS′) including, for each of a plurality of segments of an audiosignal: sinusoidal codes (CS), pulse train parameters (L₀), and firstfilter parameters (Ps); and a synthesizer (SiS) arranged to employ saidsinusoidal codes to synthesize respective sinusoidal components of theaudio signal; means (PTG) for generating an excitation signal from saidpulse train parameters (L₀); means for imposing (SEG) a spectralenvelope according to said first filter parameters (Ps) on a firstsignal (r₂′) a component of which comprises said excitation signal, andan adder for adding said synthesized sinusoidal components and saidspectrally filtered signal to produce a synthesized audio signal({circumflex over (x)}).
 20. Audio system comprising an audio coder asclaimed in claim
 18. 21. Audio stream (AS) comprising sinusoidal codes(Cs) corresponding to respective sinusoidal components of an audiosignal (x); first filter parameters (Ps) for a filter which has afrequency response approximating a frequency spectrum of a firstresidual signal, said first residual signal corresponding to said audiosignal with a signal corresponding to said sinusoidal componentssubtracted; and pulse train parameters (L₀) modelled from a component(r₂,r₃) of a second residual signal, said second residual signalcorresponding to first residual signal with a signal corresponding tosaid first filter parameters subtracted.
 22. Storage medium on which anaudio stream (AS) as claimed in claim 21 has been stored.